> Reverberation suppression?
Basically, it means that if you are in a room with lots of echo (long
decay), I can reduce it a bit.
> I guess this would help reduce local source echoes? I've never
> _noticed_ that to be a problem in my use, but I would imagine that
> using a notebook's built-in microphone, you'd get some echo off of the
> screen and stuff [also from the whole room]..
>> Most of these echoes aren't so bad, but I guess they might make the
> encoding job harder. I'd sure rather see the echo cancellation
> finished [not that I have any say on what you work on!!!].
Well, I'm still looking for help :)
> Here's the numbers I got doing vad on 655 seconds of audio (about half
> is speech, half is absolute silence [0's]).
> P3-600: 25 seconds
> Athlon XP 1700+ (1.45Ghz): 5 seconds
> P4 2.8Ghz: 8.8 seconds.
These numbers sound like a problem I has a while ago with the decoder.
The VAD shouldn't take much CPU so I suspect there might be floating
point underflows in some part, slowing down the Intel CPUs a lot (for
some reason, the AMD CPUs seem to handle underflows faster).
> Anyway, I think I might need to find a less computationally intensive
> VAD solution for the conference. VAD is currently only used when
> people connect via the PSTN, so they presumably have a decent SNR, and
> I may be able to get away with an energy envelope type of thing,
> without needing frequency domain analysis. But before I go and start
> coding this, is there any simple optimizations that can be done to the
> preprocessor when it is being used only for the VAD decision?
Have you tried using the (less accurate) VAD that's in the codec itself
(SPEEX_SET_VAD)?
Jean-Marc
--
Jean-Marc Valin
http://www.xiph.org/~jm/
LABORIUS
Université de Sherbrooke, Québec, Canada
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